I set up a new FreePBX about two months ago. I initially had issues with PJSIP extensions, and since the PBX needed to be up fast, I originally set up all the extensions as chan_sip and now that we’re up and running, I’m trying to transfer them to PJSIP.
While working on that, a few problems appeared. I don’t think they’re related to the chan_sip/PJSIP point, but I’m just mentioning that in case, well, they are.
Issue number one was an IVR missing a digit.
When a caller calls in, they can press # to enter a direct extension. A few extensions, however, would give “We have not received a valid response …”.
After turning on DTMF debug, I noticed that the PBX kept missing a digit - always the same digit for each extension.
It seems that FreePBX always misses the fourth digit of any extension ending in 1 (all extension are four digits). If the digit is dialled again, there is no issue.
There is also no issue when I try dialling into the IVR from a local extension - the issue is only on inbound calls.
Issue number two turned up this morning. All our softphones use TCP. This has been working fine, but now TCP softphones have 1) one way audio issues, and 2) hang up after approximately ten seconds, with
WARNING: chan_sip.c:4126 retrans_pkt: Retransmission timeout reached on transmission 914B3FBCFE27695518137FDCBDF88B25F4C465B5 for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions WARNING: chan_sip.c:4150 retrans_pkt: Hanging up call 914B3FBCFE27695518137FDCBDF88B25F4C465B5 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
And then send a BYE with:
X-Asterisk-HangupCause: No user responding X-Asterisk-HangupCauseCode: 18
When I switch the softphones to UDP, we have no issues.
The problem is the same with both chan_sip and PJSIP extensions.
I get the same issues whether the phones are on the local network or remote.
We’re running 188.8.131.52 with all modules up to date.
Could anyone help me with this?
Thank you in advance!